mirror of https://github.com/asterisk/asterisk
Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8554 65c4cc65-6c06-0410-ace0-fbb531ad65f31.4
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293d88108f
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0ba27e0a6b
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